Customized IVR Implementation Using Voicexml on SIP (Voip) Communication Platform

International Journal of Modern Engineering Research (IJMER)
www.ijmer.com Vol. 2, Issue. 6, Nov.-Dec. 2012 pp-4239-4243 ISSN: 2249-6645

Customized IVR Implementation Using Voicexml on SIP (Voip)
Communication Platform
Kranti Kumar Appari1
(ECE Department, JNT University, India)

ABSTRACT : An innovative application for among the internal telephones of a private organization
communication platform based on SIP (VoIP) protocol is usually a business and also connect them to the public
presented in this paper. switched telephone network (PSTN) via trunk lines.
The Voice Server is an Open-standards-based IP Communication solutions offer migration at an
voice services framework that interprets the VoiceXML organization's preferred pace. By integrating with most of
dialog markup language. It is designed to serve as a the major legacy PBXs and voicemail systems, as well as
VoiceXML interpreter implementation for VoIP platform. other business applications, most leading IP players
Although it is perfectly suitable for PC desktop empower customers to migrate to full IP based on their
applications, it can be integrated with any telephony business needs, instead of being forced to adopt
platform, messaging suite or communications solution technologies due to limitations like interoperability of the
intended to implement the VoiceXML functionality to various business applications. Successful customer
execute feature rich voice enabled applications like: auto- migration to IPPBX communications is as much about
attendant, email-by-phone, voice dialing, message processes as it is about technologies. Understanding this,
notification and reminders, contact book look-up, business leading industry players have developed detailed plans and
transaction enablement, customer relationship management processes that make migration smoother, faster, and easier
and utility applications (driving directions, flight tracking, for companies of all sizes.
audio newsmagazines, prescription refilling). In the IP communications world, telephony is just
It also allows input via speech recognition (SR) or one of the services in the network. And, this service is
"touch tone" DTMF and dialog prompting via synthesized available from anywhere in the network, independent of
speech (TTS) or recorded audio playback. And the location. For example, a multisite business may deploy the
experience with the platform shows that, it could be widely call control (IP PBX) software only at the central site, then
utilized in enterprises, groups and organizations with low- enable the remote sites to access the service remotely over
cost because of those improvements. the network.

Keywords: Voice Over Internet Protocol (VoIP), Session II. SYSTEM ANALYSIS
Initiation Protocol (SIP), VoiceXML, Text-To-Speech (TTS), 2.1 Existing Systems
Dual-tone multi-frequency (DTMF), IPPBX (Internet Till date different kinds of browsers are being used
to browse the web content like Internet explorer, Mozilla,
Protocol Private Branch Exchange)
Netscape etc. which needs a typical computer and network
connectivity to the web content. A web browser is a
I. INTRODUCTION
Software Application which enables a user to display and
A browser is a client application program that interact with text, images, videos, music and other
takes one or more input streams on a platform and executes
information typically located on a Web Page at a website on
an application that lives on one or more document servers
the World Wide Web or a Local Area Network. Text and
by interpreting markup. In the case of VoiceXML, the
images on a Web page can contain hyperlinks to other Web
application consists of the call flow logic, the prompts for
pages at the same or different website.
the application, and any associated grammars, the document
server executes portions of the application dialog by
2.2 Proposed System
delivering VoiceXML markup to the browser in response to
The existing internet protocol network is
a document request. The markup interpreter renders the
connecting VOIP (Voice over Internet Protocol) together
VoiceXML markup within an interpreter context, perhaps
enabling users to make a call in a hassle free environment.
changing the context, and then makes calls into the
Just a stable internet connection is required and a user can
implementation platform. The implementation platform make a call from anywhere in the world independent of the
contains all of the resources needed by the markup location. Proprietary systems are easy to outgrow: Adding
interpreter to render the dialog. This application deployed in
more phone lines or extensions often requires expensive
IPPBX.
hardware modules. In some cases an entirely new phone
A private branch exchange (PBX) is a telephone
system is required. Not so with an IP PBX: a standard
routing system that directs all calls from outside lines and
computer can easily handle a large number of phone lines
routes them to the appropriate phone. This type of system is
and extensions just add more phones to your network to
most commonly used in an office space. PBXs make
expand.
connections
2.3 Feasibility Study
a) Operation Feasibility:

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It is much easier to operate unlike the traditional web switching equipment, protocols, and links are already being
browsers. This project has been proposed in a user-friendly put into place. A transition network is currently in place that
environment where the hardware requirement is very less. joins the packet data world with the circuit-switched world.
No proprietary software is required. Integrated access solutions are being installed that support
b) Technical Feasibility: integrated data, voice, and other media into the Internet or
VoiceXML scripting is much easier to design. Any layman the PSTN.
can understand VXML code. The hardware complexity is Voice over Internet Protocol (VoIP) is a protocol
very less. All this project requires is a stable Internet optimized for the transmission of voice through the Internet
connection, a processor with 1GB of RAM. or other packet switched networks. VoIP is often used
abstractly to refer to the actual transmission of voice (rather
c) Cost Feasibility: than the protocol implementing it). VoIP is also known as
This project has one of the major benefits i.e. that most of IP Telephony, Internet telephony, Broadband telephony,
the software that are being used here all available for Open Broadband Phone and Voice over Broadband. "VoIP" is
Sources. The set up cost is very less. Just a stable Internet pronounced voyp.
connection is required and a user can access it from Despite a number of technological issues, real-time
anywhere in the world independent of the location. multimedia transmission (voice and video) over IP networks
and the Internet has largely been worked out. Advanced
2.4 Model Used compression techniques have reduced voice data transfer
When the user places a call to the designated rates from 64 Kbits/sec to as little as 6 Kbits/sec. Voice over
VoiceXML extension the call gets handed over to the pre- IP or VoIP can potentially allow users to call worldwide at
recorded hunt group. The particular code for a call flow gets no charge (except for the fee paid to service providers for
activated when a call is placed. After the session is Internet access). A user's IP address basically becomes a
established successfully Asterisk gateway interface invokes phone number. Additionally, computer-based phone
the VoiceXML browser and calls the initial VoiceXML systems can be linked to servers that run a variety of
script hosted on the same server. Voice Glue starts interesting telephony applications, including PBX services
interpreting the VoiceXML code for audio output and the and voice messaging.
user input by way of DTMF signals or voice commands. One of the best reasons to support packet
telephony can be seen in the service limitations of the
traditional telephone system. The switches are mostly
proprietary with embedded call control functions and
service logic. That makes it difficult to add new services. In
addition, the end devices-telephones-are limited in
functionality to a 12-key pad! In contrast, new services are
easy to add in the IP telephony world because users simply
add new telephony applications on their computers and
communicate with other users who are running the same
telephony applications.

Fig 1: Text-To-Speech (TTS)

User can control the navigation of VoiceXML code with
user’s commands. Once the user is finished with his
operation on VoiceXML the control is handed over to
Asterisk (IPPBX). Asterisk comes as a part of Fig 2: VoIP Architecture
preconfigured code terminates the call after the control is
handed over from Voice Glue. 3.2 Session Initiation Protocol (SIP)
VoiceXML is a key used to transfer text to speech There are many applications of the Internet that
or database entries to speech in a very flexible way. Any require the creation and management of a session, where a
text with its index of contents can be transferred from the session is considered an exchange of data between an
document file into forms and menus of VoiceXML files that association of participants. The implementation of these
can be read out by text-to-speech synthesis tools like Web applications is complicated by the practices of participants:
Sphere. users may move between endpoints, they may be
addressable by multiple names, and they may communicate
III. TECHNOLOGY OVERVIEW in several different media sometimes simultaneously.
3.1 Voice Over Internet Protocol (VoIP) Numerous protocols have been authored that carry various
In just a few years, the old circuit-switched voice- forms of real-time multimedia session data such as voice,
centric communications network will give way to a data- video, or text messages.
centric, packet-oriented network that seamlessly supports The Session Initiation Protocol (SIP) works in
data, voice, and video with a high quality of service. The concert with these protocols by enabling Internet endpoints


(called user agents) to discover one another and to agree on also switch calls between a VoIP user and a traditional
a characterization of a session they would like to share. For telephone user, or between two traditional telephone users
locating prospective session participants, and for other much like a conventional PBX does. The IP PBX is also
functions, SIP enables the creation of an infrastructure of able to connect to traditional PSTN lines via an optional
network hosts (called proxy servers) to which user agents gateway so upgrading day to day business communication
can send registrations, invitations to sessions, and other to this most advanced voice and data network
requests. SIP is an agile, general-purpose tool for creating, Internet protocol private branch exchange (IP
modifying, and terminating sessions that works PBX) market offers a ray of hope in the otherwise depressed
independently of underlying transport protocols and without European telecommunications industry. Encouraging
dependency on the type of session that is being established. developments in this market have seen enterprises
SIP is generic protocol for every IP capable access beginning to replace their time division multiplexing
networks. (TDM) voice networks with IP enabled/converged voice
The Session Initiation Protocol (SIP) is an data networks.
application-layer control (signaling) protocol for creating,
modifying, and terminating sessions with one or more
participants. It can be used to create two-party, multiparty,
or multicast sessions that include Internet telephone calls,
multimedia distribution, and multimedia conferences. SIP is
designed to be independent of the underlying transport
layer; it can run on TCP, UDP. It was originally designed
by Henning Schulzrinne (Columbia University) and Mark
Handley (UCL) starting in 1996. It is a 3GPP (Third
Generation Partnership Project) signaling protocol. It is one
Fig 4: IPPBX Architecture
of the major signaling protocols used in Voice over IP
(VoIP).
An IP PBX or IP Telephone System consists of
SIP handles the signaling part of a communication session.
one or more SIP phones, an IP PBX server and optionally a
VOIP Gateway to connect to existing PSTN lines. The IP
PBX server functions in a similar manner to a proxy server:
SIP clients, being either soft phones or hardware-based
phones, register with the IP PBX server, and when they
wish to make a call they ask the IP PBX to establish the
connection. The IP PBX has a directory of all phones/users
and their corresponding SIP address and thus is able to
connect an internal call or route an external call via either a
Fig 3: SIP Trapezoid Architecture VOIP gateway or a VOIP service provider.

SIP handles the signaling part of a communication 3.4 VoiceXML
session. It serves as a carrier for the Session Description The Extensible Markup Language (XML) is a
Protocol (SDP). SDP handles the media portion of the general-purpose markup language. It is classified as an
session. The transmission of voice and video content are extensible language because it allows its users to define
done by the Real-time Transport Protocol (RTP). A SIP their own elements. Its primary purpose is to facilitate the
session thus involves packet streams of RTP. SIP is a part of sharing of structured data across different information
the protocols involved in a multimedia session. The latest systems, particularly via the Internet, and it is used both to
version of the specification is RFC 3261 from the IETF SIP encode documents and to serialize data.
Working Group. In 1998, W3C hosted a conference on voice
browsers. By this time, AT&T and Lucent had different
3.3 Internet Protocol Private Branch Exchange (IPPBX) variants of their original PML, while Motorola had
An IP PBX or VOIP phone system replaces a developed VoxML, and IBM was developing its own
traditional PBX or phone system and gives employees an SpeechML. Many other attendees at the conference were
extension number, the ability to conference, transfer and also developing similar languages for dialog design; for
dial other colleagues. All calls are sent via data packets over example, such as HP's TalkML and Pipe Beach’s
a data network instead of the traditional phone network. VoiceHTML.
An IP PBX is a complete telephony system that The Voice XML Forum was then formed by
provides telephone calls over IP data networks. Typically an AT&T, IBM, Lucent, and Motorola to pool their efforts.
IP PBX system is a piece of software running on a server. The goal of the Voice XML Forum was to define a standard
Depending on the workload, that server can also be dialog design language that developers could use to build
performing other tasks, but usually it is dedicated and also conversational applications.
acts as the VoIP system's connection to the internet. In 2000, the Voice XML Forum released Voice
An IP PBX is a telephone switching system inside XML 1.0 to the public. Shortly thereafter, Voice XML 1.0
an enterprise that switches calls between Voice over IP was submitted to the W3C as the basis for the creation of a
(VoIP) users on local lines and lets all users share a certain new international standard. Voice XML 2.0 is the result of
number of external telephone lines. The typical IP PBX can

this work based on input from W3C Member companies, the VoiceXML interpreter conducts the dialog after answer.
other W3C Working Groups, and the public. The implementation platform generates events in response
Voice XML is designed for creating audio dialogs to user actions ( e.g. spoken or character input received,
that feature synthesized speech, digitized audio, recognition disconnect ) and system events (e.g. timer expiration).
of spoken and DTMF key input, recording of spoken input, Some of these events are acted upon by the
telephony, and mixed initiative conversations. Its major VoiceXML interpreter itself, as specified by the VoiceXML
goal is to bring the advantages of Web-based development document, while others are acted upon by the VoiceXML
and content delivery to interactive voice response interpreter context.
applications. A common architecture is to deploy banks of The language describes the human-machine interaction
voice browsers attached to the Public Switched Telephone provided by voice response systems, which includes:
Network (PSTN) so that users can use a telephone to 1) Output of synthesized speech (text-to-speech)
interact with voice application. 2) Recognition of spoken input
Here is a short example of Voice XML. This is a Hello 3) Recognition of DTMF input
World example: 4) Recording of spoken input
<?xml version="1.0"?> 5) Control of dialog flow
<vxml 6) Telephony features such as call transfer and disconnect
version="2.0"xmlns="http://www.w3.org/2001/vxml"> The language provides means for collecting
<form> character and/or spoken input, assigning the input results to
<block>Hello World!</block> document-defined request variables, and making decisions
</form> that affect the interpretation of documents written in the
</vxml> language. A document may be linked to other documents
The top-level element is <vxml>, which is mainly through Universal Resource Identifiers (URIs).
a container for dialogs. There are two types of dialogs: VoiceXML has become a standard and has the following
forms and menus. Forms present information and gather advantages:
input; menus offer choices of what to do next. This 1) Reduces development costs
example has a single form, which contains a block that 2) Separation between dialogue system components
synthesizes and presents "Hello World!" to the user. Since 3) Portability of application
the form does not specify a successor dialog, the 4) Re-use of Internet infrastructure
conversation ends. 5) VoiceXML is becoming a standard
The architectural model assumed by this document 6) Reduces dialogue system development time
has the following components. A document server (e.g. a 7) Additional functions can be implemented
Web server) processes requests from a client application, 8) Can develop own dialogue system with free VoiceXML
the VoiceXML Interpreter, through the VoiceXML browsers
interpreter context. The server produces VoiceXML IV. SYSTEM DESIGN
documents in reply, which are processed by the VoiceXML 4.1 UML Diagrams:
interpreter. The VoiceXML interpreter context may monitor
user inputs in parallel with the VoiceXML interpreter.

Fig 5: VoiceXML Architecture

For example, one VoiceXML interpreter context
may always listen for a special escape phrase that takes the
user to a high-level personal assistant, and another may
listen for escape phrases that alter user preferences like
volume or text-to-speech characteristics.
The implementation platform is controlled by the
VoiceXML interpreter context and by the VoiceXML
interpreter. For instance, in an interactive voice response
application, the VoiceXML interpreter context may be
responsible for detecting an incoming call, acquiring the
initial VoiceXML document, and answering the call, while Fig 6: Creating Audio Database


VI. CONCLUSION
In this paper, we proposed the Customized IVR
implementation using VoiceXML on VoIP platform.
VoiceXML provides flexibility at robust technology which
delivers its output in form of audio which will be very much
useful to all kinds of people starting from the very busy
business man to layman who doesn’t even know how to
read.
VoiceXML enables all its robust features in lines
with existing coding methodology used in today’s very
popular HTML and XML. It makes the development of
VoiceXML very easier and adoptable to presently existing
system.

References
[1] Benbin Chen, Wenchao Zhou, Yiyang Li ,Donghui
Guo, Innovative application of SIP protocol for
communication platform , Anti-Counterfeiting
Security and Identification in Communication (ASID),
2010 International Conference, IEEE,2010
[2] Larson, J.A, W3C Speech Interface Languages:
VoiceXML, Vol 24, IEEE, 2007
[3] Jianfeng Zhu, Zhuang Li, Yuchun Ma, Yulin Huang,
Realization of Extended Functions of SIP-Based IP-
PBX, Vol 3, IEEE, 2010
[4] Prasad, J.K., Kumar, B.A, Analysis of SIP and
realization of advanced IP-PBX features , Vol 6,
IEEE, 2011
[5] Gokhale, S.S , Jijun Lu, Signaling performance of SIP
based VoIP: a measurement-based approach , Vol 2,
Fig 7: Flow of Execution IEEE, 2005

V. TESTING AND IMPLEMENTATION
Test 1:
Placing two concurrent call on the vxml system

Test 2:
Checking the code how when wrong option selected

5.1 Test Execution:
Case 1:
Call from two extensions 101,102 simultaneously

Case 2.1:
After the welcome message select ECE by pressing choice 3

CASE 2.2
After selecting ECE we choose first year by pressing 1.

5.2 Result
Case 1:
Two calls successfully able to hear the welcome message on
both the calls

Case 2.1:
Able to hear the welcome message of ECE department and
prompts to select year

Case 2.2:
The calls suppose to go to the year level menu but call get
routed Thank you and it got hang up.


Customized IVR Implementation Using Voicexml on SIP (Voip) Communication Platform

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